E103IP USER MANUAL VERSION: V1.
CONTENTS 1. INTRODUCTION ..........................................................................................5 1.1 OVERVIEW OF HARDWARE ..................................................................................................... 5 1.2 OVERVIEW OF SOFTWARE ...................................................................................................... 5 2. E103 IP PHONE KEYBOARD..................................................................7 2.1 KEYBOARDS FUNCTIONS .................
.5.8 Call Service Settings.............................................................................................. 25 3.5.9 Memory Key ............................................................................................................. 26 3.5.10 MMI Filter................................................................................................................ 26 3.5.11 DSP..................................................................................................................
Function 1. Provide a Backup SIP Server 2. Support NAT, Firewall 3. Support DHCP assigning IP address, etc automatically 4. Support PPPoE (used while connecting ADSL，cable modem) 5. It can update the program through HTTP, FTP and TFTP 6. Check the dynamic voice; Soft the noise; Buffer technique of voice 7. Hold Function 8. Hotline Function 9. Speed-dial 10. Call-forward, Three-way conference call 11. DND (Do Not Disturb), Black List, Limit List 12. Auto-answer. 13. Set through standard Web Browser 14.
1. Introduction This is the user manual of E103 IP . Some configuration should be done before use the E103 IP phone, and then it can work normally. This manual will illustrate how to set the phone through keyboard and web service. 1.1 Overview of Hardware 1.1.1 The two RJ-45 network interface support the 10/100M Ethernet. The default WAN interface is a DHCP Client server. Users connect the WAN interface to ADSL or switch, and connect the LAN interface to the computer.
DTMF Firmware Upgrade l DTMF RELAY l TFTP l DTMF RFC 2833 l HTTP l DTMF SIP Info l FTP SIP Server Provide a Backup SIP Server 6
2. E103-IP phone keyboard 2.1 Keyboards functions 2.1.
Redial Dialing Volume + Call Increase the volume Volume - Call Reduce the volume FNC Off-Hook 1 Dialing On-Hook Redial the number of last time Quickly cut off the current line and off-hook again.
light begins flash at the same time, it means enable the handset register mode, the “ON/OFF” instruction light flashes at the same time, if the handset register to the base is successful, then the “charge” instruction light on the base and the “ON/OFF” instruction light on the handset will stop flickering, and will give the prompt tone for successful registration. Note: Each base phone can most register 5 handsets. 2.2.
2.2.8 Line Switch 2.2.81 Handset Line Switch 1）Suppose the line1/2 of handset is talking, the line2/1 is holding, if you want to implement line switch, first you need to press down “Hold ”, then press down “ON/OFF” of the choice line. You can implement line switch. 2) Suppose the line1/2 of handset is holding, the line2/1 is dialing, if you want to switch to line1/2 talking, first hang up line2/1, then press down “ON/OFF” of Line1/2 or press “HOLD”. You can implement line switch. 2.2.8.
3) PPPOE ◆User name ◆Password 4) QoS 2.3.3 Call Feature 23.3.1 Phone-number 1) Public SIP 2) Private SIP 2.3.3.2 Limit-List 1) Current 2) ADD 3) DEL 2.3.3.3 Black-List 1) Current 2) ADD 3) DEL 2.3.3.4 FastCall 2.3.3.5 Three Talk 2.3.3.6 Call-Waiting 2.3.3.7 Call-Forward 1) Condition 2) SIP ◆Transfer Num ◆Transfer IP ◆Port 2.3.3.8 Dial-Rule 1) End with “#” 2) Fixed Length ◆Switch ◆Length 2.3.4 SIP 2.3.4.1 Reg Status 1) Public Reg 2) Private Reg 2.3.4.1 Reg Switch 1) Public 2) Private 2.3.4.
1) Public 2) Private 2.3.4.5 Detect-server 2.3.4.6 Dtmf-mode 2.3.4.7 Interval-time 2.3.4.8 Swap-server 2.3.4.9 RFC-version 2.3.4.10 Signal-Port 2.3.4.11 Stun 1) Switch 2) Addr 3) Port 4) Expire Time 2.3.5 DSP 2.3.5.1 Codec 2.3.5.2 Handdown-time 2.3.5.3 Dtfm-Volume 2.3.5.4 Input-Volume 2.3.5.5 Output-Volume 2.3.6 System 2.3.6.1 Save 2.3.6.2 Reboot 2.3.6.3 Set Default 2.3.7 Other Setting 2.3.7.1 Syslog 1) Switch 2) Server-IP 3) Server-Port 2.3.8 Setting catalog 1.
3. through web browser to set phone Plug one end of the network line to the network card port of the computer, the other end to Lan port of the phone, phone will obtain the IP address automatically, open IE, input the IP address on Address column, then enter into the Web Setting Page. The method of obtaining the dynamic IP address is: Under the on-hook status, press “**47#”, then phone will broadcast the current IP address. 3.1 Logon The default user name and password are admin/admin and guest/guest.
3.3 Network 3.3.1 Wan Config WAN port network setting page. Support static IP, dynamic obtain IP and PPPoE. Net Traffic Timeout: when wan port network emergence failure, the phone cannot obtain IP address; the phone will auto reboot after the setting time. The default time is 2mins. 802.1x Setting: when you enable the 802.1x, after the authentication completes successfully, it should then get IP using DHCP. Username: 802.1x server username is voip. Password: the default password is 123456. Enable 802.
----Set E103-IP’s IP address in the IP Address; ----Set netmask in the Netmask field; ----Set router IP address in the Gateway; ----DNS Domain: ----Set local DNS server in the Preferred DNS and the Alternate DNS。 Configure to dynamic obtain IP ----Enable DHCP; If there is DHCP server in your local network, E103 IP will automatically obtain WAN port network information from your DHCP server. Configure PPPoE: ----Enable PPPoE ----PPPoE server: Enter “ANY” if no specified from your ITSP.
3.4 VoIP 3.4.1 SIP Config Setting page of public SIP server: Register Server Addr: Register address of public SIP server; Register Server Port: Register port of public SIP server，default port is 5060; Register Username: Username of your SIP account (Always the same as the phone number); Register Password: Password of your SIP account.
Proxy Username: proxy server username; Proxy Password: proxy server password; Domain Realm: SIP domain, enter the SIP domain if any, otherwise E103-IP will use the proxy server address as SIP domain. Local SIP port: Local SIP register port, default 5060; Phone Number: Phone number of your SIP account; Enable Register: Enable/Disable SIP register.E103 IP won’t send register info to SIP server if disable register.
No answer：If no answer, it will forward to the appointed phone; Always：The caller always forward to the appointed phone. Forward Photo Number：Call the forwarded phone number. Signal Key：Setting Signal Key; In order to prevent blocking, cooperate platform to encrypt signal, input key here, Media Key: Setting Media Key; Subscribe Expire Time: Config the time of sending subscription message; Each interval time, sending a subscription message. Mainly subscribe other’s state or voice message.
Default is RFC 3261. Park Mode: the default park mode is default; it means that the phone disenables call park function. If you want to enable call park function, choose”park1” mode, click “APPLY” button。 3.5 Advance 3.5.1 DHCP Server DHCP server manage page. User may trace and modify DHCP server information in this page. DHCP Lease Table：display the IP－MAC corresponding table that the server distributed. Lease Table Name: Lease table name. Start IP: Start IP of lease table. End IP: End IP of lease table.
Notice: This setting won’t take effect unless you save the config and reboot the device 3.5.2 NAT Advance NAT setting. Maximum 10 items for TCP and UDP port mapping. IPSec ALG: Enable/Disable IPSec ALG; FTP ALG: Enable/Disable FTP ALG; PPTP ALG: Enable/Disable PPTP ALG; Transfer Type: Transfer type using port mapping. Inside IP: LAN device IP for port mapping. Inside Port: LAN device port for port mapping. Outside Port: WAN port for port mapping.
3.5.3 STUN This page is used to set the private sip server, stun server, and back up sip server information. STUN Server setting: SIP STUN is used to realize SIP penetrates through NAT, when the phone configures IP and port of STUN server (default is 3478) and select Enable SIP Stun, common SIP server can be used to realize the phone to penetrate through NAT.
port to enhance system’s security. When this port is changed, please use http://xxx.xxx.xxx.xxx:xxxx/ to reconnect. Telnet Port: configure telnet transfer port, default is 23. RTP Initial Port: RTP initial port. RTP Port Quantity: Maximum RTP port quantity, default is 200 Notice: Settings in this page won’t take effect unless save and reboot the device. If you need to change telnet port or HTTP port, please use the port greater than 1024, because ports under 1024 is system remain ports.
Input/Output: Specify current adding rule is input rule or output rule. Deny/Permit: Specify current adding rule is deny rule or permit rule. Protocol Type: Protocol using in this rule: TCP/IP/ICMP/UDP. Port Range: Port range of this rule. Src Addr: Source address. It can be a specific IP address or network address. Dest Addr: Destination address. It can be a specific IP address or network address. Src Mask: Source address mask. Indicate the source is dedicate IP if set to 255.255.255.255.
0x28,0x30,0x38,0x48,0x50,0x58,0x68,0x70,0x78,0x88,0x90,0x98,0xb8.defa ult is 0xb8 ,oxb8 stands for best fast transmission; 28-38 is guarantee for the transmission priority for the 1st rank , 48-58 is guarantee for the transmission priority for the 2nd rank, 68-78 is guarantee for the transmission priority for the 3rd rank, 88-98 is guarantee for the transmission priority for the 4th rank. 802.1P Priority: The priority of 802.1p 3.5.
immediately. 955xx：5 digits numbers begin with 9 will be sent immediately. 10060：Number 10060 will be sent will be immediately 22xxxxxT1：7 digits numbers begin with 22 will be sent after one second 39[3,9]xxxx ： 7 digits numbers begin with 393 or 399 will be sent immediately. 3.5.8 Call Service Settings User configure the value add service such as hotline, call forward, call transfer, call waiting, 3-way conference call, auto-answer, etc in this page。 Hotline: configure hotline number.
Enable Call Transfer: Please refer to Value_add_service for detail. Enable Call Waiting: Enable/disable Call Waiting Enable Three Way Call: Please refer to Value_add_service for detail. Accept Any Call: If this option is disabled, E103 IP refuses the incoming call when the called number is different from E103 IP’s phone number. No Answer Time: no answer call forward time setting. Black List: incoming call in these phone numbers will be refused.
3.5.11 DSP This page mainly completes voice configuration. CODEC: select the prefer CODEC; support ulaw, alaw, G729 and G7231 5.3/6.3 Signal Standard: Signal standard for different area. Handdown Time: Hand down detects time. Input Volume: Handset input volume. Output Volume: Handset output volume. Handfree Volume: Hand free volume G729 Payload Length: G729 payload length VAD: Enable/disable Voice Activity Detection 3.5.
to the terminal. If there is a IP address shown on terminal (except for 0.0.0.
3.7 Config Manage Save Config: Save current settings. Clear Config: Restore to default settings. Backup Config: Backup the config file, via point the right key of mouse-à save target as….-àwill pop a save window, then type the config file name in the File name (the file type is text file) Update Configuration: Update the current configuration through configuration files.
3.8 Update Firmware 3.8.1 Update Web Update： Update the application or configuration files of the phone. The application document is .z format, and the configuration file is .cfg format. Through clicking on the "browse" button to open the upgrade file or configuration file, then click on "Update" button. After the upgrade, the phone will automatically restart. Notice：When upgrading, WEB page cannot be closed.
3.8.2 Auto Update Current Version: the system will display the current version number Server Address: FTP/TFTP server address Username: FTP server user name Password: FTP server password Config File Name: The name of configuration file Config Encrypt Key: The encrypt key of confirmation file Protocol Type: The protocol type that used for upgrading Update Interval Time: The interval time that the terminals search for new configuration file.
3.9 System Manage 3.9.1 Account Manage Set web access account or keypad password of E103 IP. 3.9.
MGR Log Level: set the MGR log level SIP Log Level: set the SIP log level IAX2 Log Level: set the IAX2 log level Please click “apply” after setting 3.9.3 Phone Book 3.9.4 Time Set This page layout is the setting of time of phone.
3.9.5 MMI SET Set the greeting information on LCD. 3.9.6 Logout & Reboot Logout：Exit the Web entry. Reboot Phone：Logout the entry, and reboot the phone. When user modifies any config of the phone, it will take effect after being rebooted, you can enter into this layout and click “Reboot”. And the phone will be rebooted automatically. Note：Reboot IP phone, some setting needs to reboot to make it works. Please always save configuration before reboot, otherwise the setting will return to previous setting.
4. Operating Method for Dialing 4.1 How to dial IP Phone You can make a call after being made a proper setting on your phone. Please confirm whether all the net wires are connected correctly. If you want to make a call, you can make it after dialing the number and then pressing “#”. You can find IP address through the menu. Modify the IP address of the computer, and making it the same net as the phone.
4.2.2 SIP setting： Enter into the VoIP à SIP Config to set the layout config and sip account information: Register Server Addr: Register address of public SIP server Register Server Port: Register port of public SIP server，default port is 5060. Register Username: Username of your SIP account (Always the same as the phone number) Register Password: Password of your SIP account.
4.3 How to use the dial rule? E103-IP provide flexible dial rule, with different dial-rule configure, user can easily implement the following function: ----Replace, delete or add prefix of the dial number. ----Make direct IP to IP call ----Place the call to different servers according the prefix. You can click “Add” to add a new dial rule. Below is the detail setting of the dial-rule: Phone Number: The Number suit for this dial rule, can be set as full match or prefix match.
Instance: 2T rule: If the call starts with 2, the first 2 will be deleted, and the rest number will be sent to private SIP server. 3T rule: If the call starts with 3, the first 3 will be deleted, and the rest number with be sent to public SIP server. 123 rule: Dial 123 and will send 06332221015 to your server. It is used as speed dial function. 0T rule: If the call begins with 0, the first 0 will be replaced by 86. It means that if you dial 06332221015 and AG-188 will send 866332221015 to your server.
MWI LED will also blink to prompt. After the voicemails are picked-up, MWI LED will stop blinking.
IC Warning: This device complies with Industry Canada license-exempt RSS standard(s). Operation is subject to the following two conditions: (1) This device may not cause interference, and (2) this device must accept any interference, including interference that may cause under operation of the device. Privacy of communications may not be ensured when using this telephone.
FCC Caution: Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment. This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.
